1. Field of the Invention
The present invention relates to a network, a server apparatus, an Internet protocol (IP) terminal device, and a speech-quality control method used in the network, the server apparatus, and the IP terminal device. Particularly, the present invention relates to control of the speech quality in a network including IP terminal devices.
2. Description of the Related Art
This type of network has heretofore been constituted by connection of an IP-PBX (Internet Protocol-Private Branch exchange) to a fixed phone terminal, a radio terminal such as a personal handy-phone system (PHS), a voice over Internet protocol (VoIP) terminal or the like via a local area network (LAN).
As the VoIP terminal, in addition to an IP phone terminal and an Internet phone terminal, there is a session initiation protocol terminal (hereinafter they are collectively called IP terminal). The SIP is a communication protocol for use in starting or ending multimedia communication such as sound communication (fixed phone, cellular phone, etc.), video communication such as television phone, chat (conversation by characters) and the like in an environment of an IP network using data having a form referred to as an IP packet.
In the IP network, in general, connection-less type communication is performed without confirming connection to a target as in electronic mails. On the other hand, in the fixed phone, in general, connection type communication is performed while confirming the connection with the target. The SIP realizes the connection type communication in the IP network.
The SIP basically comprises methods (operations) such as INVITE (session between users is established), ACK (acknowledgment), CANCEL (INVITE is ended during the establishment of the session), and BYE (the end of the session). The respective methods are exchanged as requests and responses to the requests between clients and servers to thereby establish or end the session.
Moreover, the SIP has characteristics that applications can be comparatively easily prepared. For example, when a new service is added to H.323 of ITU-T for use in the IP phone, an H.450.x protocol which defines the H.323 additional service is added, and all H.323 end points on the network and software of a gate keeper need to be updated. However, in the SIP, an SIP application server which provides the new service is added, and the corresponding application is added. Then, the new service is usable.
In the SIP described above, an “offer/answer model” using a session description protocol (SDP) is defined. In this “offer/answer model”, the SDP is used in the bodies of “INVITE” and “200 OK” corresponding to the “INVITE” in order to negotiate the media information used in a session. The “INVITE” denotes a method used for establishing a session between subscribers and the “200 OK” denotes a success response.
In the negotiation of session information in the “offer/answer model”, an offerer who wants to establish the session transmits the session information (hereinafter referred to as an offer), which is represented in the SDP and which the offerer wants to use in the session, to an answerer. The offer includes the IP address, the port number, and the types of the medium and the CODEC. The offer is described in the body of the SIP message.
In response to the offer, the answerer transmits the session information (hereinafter referred to as an answer), which the answerer wants to use in the session, to the offerer. The answer is also described in the body of the SIP message. In this manner, the negotiation of the session information is completed between the offerer and the answerer.
FIG. 9 is a sequence chart showing a known bandwidth control operation. Specifically, when the bandwidth control is to be performed between IP terminals (for example, the Sip corresponding terminals described above), in e1 in FIG. 9, an Sip corresponding terminal #1 transmits “INVITE (w/SDP)” including the offer (the bandwidth information) to an Sip corresponding terminal #2. In e2, the Sip corresponding terminal #2 returns “180 Ringing” to the Sip corresponding terminal.
In e3, the Sip corresponding terminal #2 determines an optimal bandwidth from the bandwidth information transmitted from the Sip corresponding terminal #1. In e4, the Sip corresponding terminal #2 returns “200 OK(w/SDP)” including the bandwidth information to the Sip corresponding terminal #1 as a response. In e5, the Sip corresponding terminal #1 returns “ACK” to the Sip corresponding terminal #2. In this manner, the session is established between the Sip corresponding terminals #1 and #2.
FIG. 10 is a sequence chart showing a known adjustment operation of an audio level. When the adjustment of an audio level is to be performed in the IP terminals (for example, the Sip corresponding terminals described above), in f1 and f2 in FIG. 10, the Sip corresponding terminals #1 and #2 respectively set audio levels. In f3, the Sip corresponding terminal #1 transmits “INVITE (w/SDP)” to the Sip corresponding terminal #2.
In f4 and f5, in response to the “INVITE (w/SDP)”, the Sip corresponding terminal #2 returns “180 Ringing” and “200 OK(w/SDP)” to the Sip corresponding terminal #1. In f6, the Sip corresponding terminal #1 returns “ACK” to the Sip corresponding terminal #2. In this manner, the session is established between the Sip corresponding terminals #1 and #2.
Japanese Unexamined Patent Publication (JP-A) No. 2000-179638, “SIP: Session Initiation Protocol” [RFC (Request For Comment) 3261, 8 to 34 pages, June 2002], and “An Offer/answer Model with the Session Description Protocol (SDP)” [RFC 3264, 1 to 25 pages, June 2002] disclose the above-described earlier techniques.
However, there are problems in that the bandwidth control can originally be performed only between IP terminals in known communication between the IP terminals and in that each IP terminal must individually adjust the audio level in known communication between IP terminals.